I just registered myself for the first time ever to a forum of anykind

But this just seemed like a place of people who know the stuff they talk about,and that's always good. Also this place has had by far the best attitude towards"the new guy" making his first post, so bear with me, since this truly is my first post

Hi.

Welcome to HA. I sincerely hope your experience here is as enlightening and meaningful as mine has been.

I fail to understand the b value of 128 on -V 0. Is there a simple explanation for why the bitrate should not vary in the full 32-320 kbps range? Also, can -b 32 override -V 0?For example would running:-V 0 -b 32 -B 320make any difference at all, or would the preset-style V switch override the b and B?

/Add: I think the simple explaination why 32-112 range should not be used at all is that more or less any samples encoded with these bitrates will not be transparent (except digital silence - which is btw encoded as 32kbps anyway).

On a resembling matter, I've been using Audiograbber (ugh!) with the lame_enc.dll (3.96.1), with -V 2 equvalent, and it appears that it uses -b 96 as default? Should be 128. It could be that Audiograbber sets -b 96 internally, but it seems unlikely. I ask because I wonder if the lame dll has the same mappings as the exe.

I fail to understand the b value of 128 on -V 0. Is there a simple explanation for why the bitrate should not vary in the full 32-320 kbps range? Also, can -b 32 override -V 0?For example would running:-V 0 -b 32 -B 320make any difference at all, or would the preset-style V switch override the b and B?

you shouldn't mess with the --presets at all. So just use -V0 and that's it. No -b or -B unless you like to waste space or worse sound quality Anyway, LAME 3.96.1 uses -b 128 as default, however the newest LAME 3.97 alpha builts use -b 32. That's why it is the best to just stick with the presets. Note: LAME Alpha and Beta release shouldn't be used for anything else than testing!

--------------------

--alt-presets are there for a reason! These other switches DO NOT work better than it, trust me on this.LAME + Joint Stereo doesn't destroy 'Stereo'

The way I understand it is this: lame has different variable bitrate modes. you have ABR and a true VBR mode. (the ABR is what was used with the old r3mix.net settings)However, apparantly the VBR mode is somewhat unreliable below 112 Kbits/sec. It will mostly produce good encodes but may output some low-quality parts. Therefore, lame in VBR mode (which is what the presets use) has a bitrate floor of 112 Kbits/sec.

I'm new to this forum. I registered because I needed some encoding information for my new-born Creative MuVo TX SE. Since it's only 256MB, I'd like to fill it up with proper encodings.

Sorry if this is the wrong section or something. I tried searching for topics about mp3 player encodings but failed.

So. I use dBPowerAmp which has command line access to Lame, and Lame is the latest 3.97b I believe. The one recommended here anyway.

Since I'm probably going to invest to Sennheiser PX-100 or something similar, I figured -V 6 might be appropriate.

-V6 --vbr-new was in mind actually. Now I have read about this -Y parameter, which cuts everything above 16kHz. I tried it, but it just produced the same results as in normal -V6. Also, I checked the full command line reference, and it was not there. Is this somehow obsolete and/or replaced by some other parameter?

I put my money on --highpass 16 , but I dont think it did anything. Probably not the switch I'm searching for or completely wrong anyway.

I'm not an Audiophile. I was considering -V8 but thought that new headphones might require the -V6 so I was sticking with that. I guess its pretty good.

I can now fit 3 -V2 albums on this player. More is required and that quality is an overkill for me anyway (although I do encoded all my CD's with that).

Thank you for your time!

And the final question is this: The whole thing with --alt preset being ABR, was this true in some versions? I always wanted the recommended --alt preset standard but ended up with bitrates never topping 224kbps. With the latest Lame though, no problems, just curious about why did I had this experience in the past.

-V6 --vbr-new was in mind actually. Now I have read about this -Y parameter, which cuts everything above 16kHz. I tried it, but it just produced the same results as in normal -V6. Also, I checked the full command line reference, and it was not there. Is this somehow obsolete and/or replaced by some other parameter?

I put my money on --highpass 16 , but I dont think it did anything. Probably not the switch I'm searching for or completely wrong anyway.

-V6 --vbr-new uses a lowpass of 15600 Hz. You should not add any other switches. In my opinion -V6 --vbr-new is a good and balanced option to use with portables. I have found that it produces bitrates from about 100 to 150 kbps with various music genres (http://www.hydrogenaudio.org/forums/index....ndpost&p=335491).

QUOTE

I'm not an Audiophile. I was considering -V8 but thought that new headphones might require the -V6 so I was sticking with that. I guess its pretty good.

In my experience -V8 is inferior. It resamples to 32 kHz and uses a lowpass of 12500 Hz, but that doesn't even solve the quality problems -V8 has below 12500 Hz. However, just the lowpass setting clearly changes the perceived quality with many music types if the listener can hear frequencies over 12500 Hz. On the other hand, hearing a lowpass of 15.600 Hz is much harder. You can try that for example by encoding a test sample with -b 320 and -b 320 --lowpass 15.6. (I have done personal ABX tests and I suppose other listening tests that can confirm this have been published at HA.)

Edit: There seems to be a small typo in the LAME documentation: "Set an highpass filtering frequency."

I tried -b 320 --lowpass 12.5 just for kicks, and the filesize was the same, however the sound was clearly inferior.

I need to mentiont that I never understood these kHz terms. Sometimes they represent how high a sound is and sometimes... not? Tried also --lowpass 15.6 and it sure was transparent to the original.

Cutting some more or less inaudible high frequencies off can make the overall audible quality better. The space is always limited with MP3 files. LAME adjusts this automatically so you don't have to worry about manual tweaking.

I mentioned -b 320 just because it is the highest quality setting LAME 3.97b1 has. It is always constant 320 kbps so the resulting file size is also constant like Gambit said. Using it for testing the audibility of a lowpass setting would minimize the possible effect of other factors.

Sorry for possible offtopic. I just found this topic as closest to my problem, but it can equally well be a new topic.

So, what's the reason for me to be confused. When I've changed my SB Live! 5.1 to Audigy 2 ZS, I just started to hear some higher frequencies of sound (including my huge collection of self-grabbed mp3s). And I used (and use LAME to encode them.

I started with YAMP encoder which was using LAME 3.80, then I moved to EAC & LAME 3.92 (newest release for that time). I've been hearing and analyzing the sound spectre of output mp3s, and arrived at a conclusion that the best (in size/quality) variant for me is:

CODE

-b 32 -m j -h -V 2 -B 320

Then there was 3.93 and 3.93.1. I've been grabbing and compressing and expanding my mp3 collection. Grabbing, and compressing, and extending...

Then here came the 3.95.Right away I noticed that mp3s I'm getting with it (using same parameters) are oftenly slightly smaller. That was the reason to REGRAB and RECOMPRESS all my collection (about 40 or 50 GB for that time).

Then there were 3.95.1, 3.96 and 3.96.1... Regrabbing, recompressing, extending collection...

First time there was some kind of euphoria - I started to hear HIGHER sounds on my computer! All that hi-hats' and cymbals' high harmonies, and so on...

Then it turned to some kind of depression. I started to hear mp3-specific distortions in MY MP3s! Which were made with my LAME parameters (I also recommended using them to all my friends... and so on). It was a real shock for me, because I used to consider my mp3s ideal.

So, I started to re-examine various mp3 codecs, incl. FhG, GoGo, SCMPX, Blade, Xing and recently appeared LAME 3.97.

The results I got for LAME shocked my once more. Awful, just AWFUL distortions all above 16-17 KHz. Even 320 at Stereo (not Joint Stereo) mode, and even with filter disabling (-k). I tried lot of different options and their combinations, all of them resulted much worse quality then trivial 3.93.1 at 320, Stereo mode (with -k), or at VBR2 Stereo (with -k also), and even worse then LAME 3.93.1 of GoGo gives me at 224 kbps or even 192 kbps (no matter Stereo or Joint Stereo mode)!

I tried using q0, and V0, and so on - it didn't improve anything enough.

I also tried various mp3 decoders: Winamp 5.1, Adobe Audition (in which I analyze sound frequency spectre) and LAME by itself. In Audiotion results were even more awful, then in Winamp and LAME, but in them both spectre was still too bad at look (at freqs above), and I heared that all in Winamp.

Even more - those distortions (looking like some kind of collapse of frequency spectre graph) grew more and more for 3.95.1 -> 3.96.1 -> 3.97. I just did't beleive my eyes that LAME produced that collapse.

So I temporary turned back to 3.93.1, which produces an ideal graph allover the sound spectre (even on freqs above 20 KHz!) when using the following parameters:

CODE

-b 128 -m s -h -V 2 -B 320 -k

The resulting mp3 freq graph doesn't drift in any significant way from the from original WAV graph. 320 kbps' graph (@3.93.1) also doesn't differ from the original in any way, but average filesize increases significantly.

Could your please explain, what happened with LAME starting from 3.95? (or 3.94, which I didn't test)Could it be some problem of my decoder (even despite I used LAME to decode mp3s)? Or what?

Thank your in advance.

P.S. I can post, or send, or attach here (if it's allowed), or put at your disposal in any other way the graphical results of spectral analysis I carried out.

-k will absolutely destroy the quality of the files. Also, M/S stereo will make things sound WORSE. Leave it as joint stereo (it is NOT lossy in LAME). -h does nothing for VBR, -q0 does nothing for VBR. Setting min/max bitrate is not needed in 3.97 when using VBR.Spectral analysis is virtually useless for measuring sound quality.

Stop screwing with the parameters and just use -V2 --vbr-newNothing more, nothing less. Try LAME 3.97 or 3.98a7 if you prefer going a little on the wild side. Then ABX it if you still think you hear "distortions".

But, as far as I understood, ABX test is based on "blind" listening to audio, but my acoustic system (and perhaps my ears ))) is not ideal, so in future (when I purchase a better one) I could regret my choise made now. Are there any less subjective, MORE objective test? Which can reveal the real (no only heared) differences between two soundtracks?

My father offered my to "subtract" original signal from the encoded->decoded one, and then to estimate the "difference signal" remained. Is this method good, how do you think? Are there any software to do that? AFAIK, Sound Forge and CoolEdit can't...

Or may be some graphic tests?

And one more question. You say spectral analysis couldn't be objective in this way. But if it shows some evident distortions, collapses etc - could they NOT be real distortions of sound and its quality? Especially if they aren't above 20 KHz, but within 16-20 KHz interval?

(Of course, I agree that even identical graphs do not obligatory mean identical sound, or sound of identical quality.)

P.S. By the way, I did't understand from the FAQ, what does -vbr-new key?

To me the main problem is that you seem to think it's absolutely necessary to get everything until 20 kHz.However our ears are less and less insensitive the more you go beyond 16 kHz. This is true even for young people though they are able to hear stuff beyond 16 kHz.

If you care so much about that you should absolutely do an abx test. You don't need good equipment to find out whether or not you need HF beyond say 18 kHz.Try Lame -V3 which has an 18 kHz lowpass and report us about whether or not you're really missing HF or find things distorted (which of course is possible only if you can abx something).

You should be a lot more afraid of -k which might give you more HF but as well a much higher chance that you get audible distortions.

Are there any less subjective, MORE objective test? Which can reveal the real (no only heared) differences between two soundtracks?

Unlike your ears, your eyes aren't meant for hearing.

QUOTE (ThyBzi @ Nov 30 2006, 10:22)

My father offered my to "subtract" original signal from the encoded->decoded one, and then to estimate the "difference signal" remained. Is that method good, how do you think?

No.

QUOTE (ThyBzi @ Nov 30 2006, 10:22)

CoolEdit can't do that...

Acutally CoolEdit can, but it tells you absolutely nothing about how your brain interprets what your ears are hearing.

QUOTE (ThyBzi @ Nov 30 2006, 10:22)

Or may be some graphic tests?

NO! Listening tests are all that matter.

QUOTE (ThyBzi @ Nov 30 2006, 10:22)

And one more question. You say spectral analysis couldn't be objective in this way. But if it shows some evident distortions, collapses etc - could they NOT be real distortions of sound and its quality?

Of course they represent real "distortions", but you need to ask yourself if you can hear these distortions in a blind listening test.

QUOTE (ThyBzi @ Nov 30 2006, 10:22)

Joint Stereo "pushes together" channels, when they spead stronger than in "regular song"... Am I wrong?

But, as far as I understood, ABX test is based on "blind" listening to audio, but my acoustic system (and perhaps my ears ))) is not ideal, so in future (when I purchase a better one) I could regret my choise made now. Are there any less subjective, MORE objective test? Which can reveal the real (no only heared) differences between two soundtracks?

My father offered my to "subtract" original signal from the encoded->decoded one, and then to estimate the "difference signal" remained. Is this method good, how do you think? Are there any software to do that? AFAIK, Sound Forge and CoolEdit can't...

CoolEdit can do that easily, but...

Our ear is extremely complex. "Visual" differences, even "extremely evident" visual differences... may not be audible. Why? Simply because our hearing, whcih comprises the ears, the inner ear system, the auditory nerves, and the brain, will perform some extremely complex post-processing and filtering.

Case in point: Try encoding with the highest quality of LAME ( -V 0), or Vorbis (-q 10). Decode. Subtract from the original wave. You will see "extremely evident" visual differences. But even if you pump the decoded wave (i.e. not original) through a 1-million-dollar universal-audiophile-grade equipment... the chances that you will hear any difference is less than 1 ppm (part per million).

QUOTE (ThyBzi @ Dec 1 2006, 01:22)

Or may be some graphic tests?

Like I said above, what's visible to your eye may never, ever be audible to your ear.

QUOTE (ThyBzi @ Dec 1 2006, 01:22)

And one more question. You say spectral analysis couldn't be objective in this way. But if it shows some evident distortions, collapses etc - could they NOT be real distortions of sound and its quality? Especially if they aren't above 20 KHz, but within 16-20 KHz interval?

Spectral analysis is not un-objective, it is simply useless. "Visible" distortion, even "extremely evident" visible distortion, like I wrote above, may not be audible to your ear even the slightest.

On higher bitrates, Joint Stereo is actually Mid-Side (MS) Coding. Instead of coding L & R channels separately, it encodes M=(L+R)/2 and S=(L-R)/2. The L & R channels can be recovered very easily: L=M+S and R=M-S. The reason for using MS is that since the S channel usually requires far less bitdepth, encoding using MS will be smaller than encoding using LR, provided that the difference between L & R are not too big. LAME is capable of determining when to use MS and when to use LR.